Hi,
FYI
It always returns 200 result = 0.
Please do help me in this regard.
Best regards
Somesh S. Shanbhag
--- somesh s [EMAIL PROTECTED] wrote:
Hi All,
I have some problem while capturing DTMF digits in
AGI
script.
My configuration for user is ..
[9009]
type=friend
Hi
I have something similar, what happends is that intermittantly,
(especially when DTMF tones are played) the call does not hang up
when the timeout expires. It looks like it is related to your issue.
Please let us know if you find any answers to this bug.
Thanks
Clive
On 18 Jul 2005 at
Hye Paul,
You were right. Only 1 of 5 attempts or sometimes 7 attempts managed
to go through the complete multi page tiff fax. This is bad.
Is there an other alternatives which can solve this?
Can HylaFax be able to fax multi page tiff file?
-eddie-
On Mon, Jul 18, 2005 at 12:03:42AM -0500, Kristian Kielhofner wrote:
trixter http://www.0xdecafbad.com wrote:
On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote:
OT:
Not a Knoppix, actually. You can't do anything useful with it without a
HD install. A while ago I needed badly to test
Any ideas anyone ?
Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___
Computers are not intelligent.
I have been beating my head against the wall trying to get my ATA186 to
send through the *8 (call pickup) sequence back to Asterisk.
The Administrator's Guide from Cisco would indicate that the first
element in the default dialplan *St4- would mean that any sequence of
digits following a *
Tzafrir Cohen wrote:
I should be releasing a much improved Live version of AstLinux
within a week or so. A test version was announced on my mailing list a
while ago, with pretty good results so far. It will be AstLinux 0.2.8, and
available as an ISO (as well as the Windows install package,
On Mon, Jul 18, 2005 at 02:40:40AM -0500, Kristian Kielhofner wrote:
I tried astlinux 0.2.6 as well, however as a live CD it wasn't useful,
because I could not write any modified configuration on the live /etc .
Thus I could not run asterisk with the modified configuration without a
proper
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
The next community meeting is
Hi,
I am looking for inexpensive isdn card that supports nt mode with asterisk.
Does it mean that if mISDN has nt mode support for a specific card that
asterisk has too? If yes, I presume I could buy any of the cards listed
on the PBX4Linux page: http://isdn.jolly.de/cards.html Or will
Hi Kape,
Life is generally good with bristuff and the quadBRI cards. However I've
got a concern: how does one return a busy signal to the telco when all B
channels are busy? Right now, when all channels are in use, the remote
caller is kept waiting until the telco times out and finally get a busy
The 7665 is not an ip phone but a proprietary digital system phone for kxtda
systems.
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Felder
Sent: 18 July 2005 04:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi Erik,
You put me to a page which refers to high load on CPU on SMP. Nothing to do
with memory leak. Furthermore I am not running SMP.
Any other suggestions in which direction to look? Am I the only one
experiencing this ?
Do you mean if I update to the today's CVS the memory leak issue will
Steve Gladden wrote:
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
The idea would be to get a network
Eddie wrote:
You were right. Only 1 of 5 attempts or sometimes 7 attempts managed
to go through the complete multi page tiff fax. This is bad.
Is there an other alternatives which can solve this?
Can HylaFax be able to fax multi page tiff file?
HylaFAX can (we ar doing this now), but not with
test
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salam if anyone uses fastagi then plz help me...
The AGISERVER starts successfully:
Jul 18, 2005 2:54:50 AM net.sf.asterisk.util.impl.JavaLoggingLog infoINFO: Thread pool started.Jul 18, 2005 2:54:51 AM net.sf.asterisk.util.impl.JavaLoggingLog infoINFO: Listening on *:4573.
but I m
On Sat, 2005-07-16 at 16:47 +0200, Zoltan Szecsei wrote:
Hi,
Can anyone please point me in a direction as to how to set up these 2
pci cards with AAH 1.1?
Rather load [EMAIL PROTECTED] 1.3 - fixes other problems
I have (am still) googling left, right center - but haven't found a
hi i am new to asterisk and i want to configure
trunk with a voice gateway as i read i must have a zaptel card installed in
order to do so. but i want to configure the trunk without any cards
installed in the server is there anyworkaround to do
this.
Hello, I've then same probleme with sip rtp packets with different
Routers. This is perhaps not a vpn problem !
Sip with french livebox France telecom - don(t work
Sip with Livebox pro - ok
Sip with Bewan adsl router - don(t work, but with last firmware, ok.
Their is a problem with ZIP Rtp
NTL install isdn 30. No idea how good they are though.
Chris
- Original Message -
From: 1 2 [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, July 07, 2005 2:50 PM
Subject: [Asterisk-Users] isdn30 / pri lines in the UK
anybody recommend a supplier in the UK for a
Ask this question on asterisk-bsd
Chris
- Original Message -
From: Jack Towards [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, July 18, 2005 4:38 AM
Subject: [Asterisk-Users] FreeBSD 5.4 (Asterisk 1.0.9) - Playback ,MP3Player
and Musiconhold not working
I
Hello;
I have already the same problem, i can't solve this. Is there anybody to
help me?
Thanks
Erdem HAKI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKİ
Sent: Friday, July 15, 2005 4:44 PM
To: 'Asterisk Users Mailing List -
hi,
i am getting unsolicited NOTIFY messages from Asterisk after the
subscription.
Is this type of NOTIFY messages is supported in any of the RFCs..?
Because, the SIP stack with 3265 compliance, does not support any such
NOTIFY messages and discarding those.
I need the justification for
Also NTL don't drop the leading 0 on incoming numbers like BT do.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton
Sent: 18 July 2005 11:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Generally speaking one works against one's own best interests when he
reminds the group that he has been posting on a topic repeatedly without
anyone answering.
Yes agreed,
In this case my only intention was to acknowledge the fact that I
realized I was asking a 3rd time and hopefully not
My 2c worth...
For the beginner, AAH is great. The PC that you install on will be
totally reformatted / fdisk-ed (assuming single drive - etc).
With AAH 1.3 - the installation goes to sleep and sort of finishes
when its Syncing with a Time Server. A reboot at this point seems to do
no harm.
As
Thanks!
Very true
I have a client who is very interested in Asterisk but they are holding off
because I have been unable to give them a solid answer if it will work
this way.
I have found bit's pieces that it *might* work with SNOM phones
and something called a 'hint application' but have
Steve:
Most likely you haven't received a solid answer because the standards
defining
this behavior are still developing. Features like monitor line and
multi-line appearance
depend upon functionality in the phone set too. Vendors are implementing
these
features in their products but in
On 18 Jul 2005, at 12:06, Lee Archer wrote:
Also NTL don't drop the leading 0 on incoming numbers like BT do.
What NTL do seems to vary somewhat. I think it depends on
the switch and datafill people. This has it's plus side, you can
sometimes get _exactly_ what you want, but it does mean
Hi,
Can you tell me how to configure Hylafax + Asterisk in order to be able
to receive/send faxes.
Best regards,
Guan
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To
Make sure, you include remote office's lan in the localnet directive
(otherwise, they'll use the wan ip address, and that may be the
problem...)
Julian.
On 7/15/05, Peter Osborne [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk for my PBX, I have a remote office that is connected by a
This function is based on a non-standardized extension to SIP made
by Broadsoft. I have all the specs and are looking into this. Don't
expect anything to happen quickly though, I have to complete another
large SIP project first (SIP Transfers) and then start looking into
this. It requires quite a
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
Sent: Monday, 18 July 2005 6:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Panasonic KX-T7665 and Asterisk?
The 7665 is not an ip phone
Features like bridged line appearance are expected to be available in
release
8 of Cisco's SIP image. I do not have an ECD for this release.
Olle E. Johansson wrote:
This function is based on a non-standardized extension to SIP made
by Broadsoft. I have all the specs and are looking into
hello perl experts
i am working with ast-rad-acc.pl from
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
i dont know why $cdr{'DNID'} and $cdr{'CALLERID'}
under 'sub send_acc {' are empty. i m successfully
connected with asterisk manager and when call i hangup
my perl
Good Day list,
Does anyone know if it is possible to setup asterisk such that
it passes DTMF Tones through from One channel to the next transparently.
I have a situation where asterisk is answering the phone on
Channel 1 (first channel of a PRI) and then bridges this call to
i get lots of the below from friday 15.7.05 cvs as well
ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ...
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
Dear all,
Does reinvite work for a SIP to SIP call if there are more than one
Asterisk between the clients? An example scenario:
A --- |Asterisk 1| --- |Asterisk 2| --- B
client A behind Asterisk 1 calls client B behind Asterisk 2, after the
connections have been established, the Asterisks
Steve Blair wrote:
Features like bridged line appearance are expected to be available in
release
8 of Cisco's SIP image. I do not have an ECD for this release.
Do you know which standard they base this on?
/O
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Hmmm.. I was unaware of the localnet parameter. I don't think the remote lan
should be added to it though, since asterisk still needs to go through a
gateway (the VPN itself) to access the remote LAN.
Pete
On 18 July 2005 7:53 am, Julian J. M. wrote:
Make sure, you include remote office's lan
Olle E. Johansson wrote:
Steve Blair wrote:
Features like bridged line appearance are expected to be available in
release
8 of Cisco's SIP image. I do not have an ECD for this release.
Do you know which standard they base this on?
Great question ;-) No I do not for obvious
Ronald Hartmann wrote:
Does anyone know if it is possible to setup asterisk such that
it passes DTMF Tones through from One channel to the next transparently.
I don't believe this is possible, no. If you are using all Zap channels
(TDM cards) and don't enable _any_ DTMF-controlled
1 2 wrote:
i get lots of the below from friday 15.7.05 cvs as well
ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ...
I will be looking into this issue later today.
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1 2 wrote:
i get lots of the below from friday 15.7.05 cvs as well
ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ...
I'm seeing these as well, via the 07-17-2005 CVS HEAD
Doug
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Astricon 2005 will take place in Anaheim, California October 12-14 2005.
Astricon is the Asterisk conference, arranged by IPsando LLC in
cooperation with Digium.
We are now looking for speakers. The conference will be bigger than last
year, so we are looking for more speakers in the conference
Mikko Suniala wrote:
client A behind Asterisk 1 calls client B behind Asterisk 2, after the
connections have been established, the Asterisks issue reinvites and
they will step out of the media path so that RTP traffic will stream
directly between the clients.
Yes. The media path will
I have a telecomFM CellRoute GSM. I want to route call to the cell
phones with that device. Has anyone experienced in connecting an
Asterisk pbx to telecomFM CellRoute GSM
successfully? Please help me to integrate this device with Asterisk Server.
Thanx in advance,
Yusuf Iqbal
I have hylafax attached to a Sipura ATA, and just have Asterisk route
calls to the fax DID to that ATA. I wouldn't recommend doing this as
it is highly unreliable and only about 50-60% of the faxes actually
finish.
As far as I know, you can't route faxes to hylafax within the same
box, however
Your question isn't very clear, but there are a vareity of trunks
available. You could get a trunk that uses the zaptel card, such as a
PRI/T1/E1. Or you could get a 'virtual' trunk that uses IP such as IAX
or SIP. There are numerous providers of IAX and SIP trunks out there,
so look around.
I agree with Steve. The ability to have
bridged extensions appear on multiple phones is a vital to many business
users. For Asterisk to seriously take on the Avaya's and Nortels and
Cisco's of the world, this is a feature that needs to be
implemented.
It is apparent that this is not
Jean-Louis curty ha scritto:
7910 works fine wiz asterisk but you can not transfer calls, for that
reason I will sell mine if somebody is interested...
chan_sccp has native transfer right now (still under development)
chan-sccp.berlios.de
Sergio
I'm trying to setup asterisk to accept call from Teliax, request the
10 digit number from user, then dial it thru the VoIPJet. If I'm not
wrong I will be charged by both providers because both connection is
active during conversation. So my question is can I set the things so
that I pay only to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: Sunday, July 17, 2005 7:50 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] SoftPhones: Bad, or just bad QoS?
On Sun, Jul 17, 2005 at 05:19:05AM -0400, Tom
Add another span= line and then extra chans .. look in the zaphfc build
directory and you will find some examples
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: 18 July 2005 12:52 AM
To: Asterisk Users Mailing List - Non-Commercial
Does anyone know if it is possible to setup asterisk such that
it passes DTMF Tones through from One channel to the next transparently.
I don't believe this is possible, no. If you are using all Zap channels
(TDM cards) and don't enable _any_ DTMF-controlled features in the
Dial()
Spoke to Klaus-Peter about this PCI performance issue. He says it has to do
with the CPU not supporting cpufreq stepping. I had to get a quad card to
get the issue resolved.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Szlezak
Sent: 17 July
Hi Guys,
I am using the latest stable version of asterisk which I updated yesterday,
but I keep getting this error message for some of my accounts. Can anyone
explain to me what does this mean? WARNING[3032]: chan_sip.c:4832
check_auth: Stale nonce received from.
Thanks
Joel
Hello,
There's this device called VoiceBlue GSM gateway.
It talks gsm on one side and SIP on the other side.
Have a look at:
http://www.voip-info.org/tiki-print.php?page=How+to+connect+VoIP+GSM+gateway+to+Asterisk+PBX
yep, but it is very expensive, I found. Even cellphone + cellsocket +
FXO
I'm trying to setup asterisk to accept call from Teliax, request the
10 digit number from user, then dial it thru the VoIPJet. If I'm not
wrong I will be charged by both providers because both connection is
active during conversation. So my question is can I set the things so
that I pay only
Use to providers for the call, pay two providers for the call.
You have two call legs so you are using two channels bridged at your *
box.
You will have to pay for those to legs...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of code
select
Sent:
as a suggestion, please play a little with the next parameters in
zapata.conf read the docs in voip-info about these parameters an may
me you will be able to fix your problem.
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=-4
immediate=yes
busydetect=yes
Hello All,
Recently, I installed TDM04B 4 FXO card on to my Asterisk box and
installation went perfect.
The only problem I am facing is the Voice mail has very poor quality
when any users leave voice message via PSTN line.
We can not hear either from the extension nor from the WAV email
Joel Jn-Francois wrote:
Hi Guys,
I am using the latest stable version of asterisk which I updated
yesterday, but I keep getting this error message for some of my
accounts. Can anyone explain to me what does this mean?
WARNING[3032]: chan_sip.c:4832 check_auth: Stale nonce received from.
Hey guys,
We have a couple of Nokia 32 GSM units left over from a client's *
installation. The units than can be hooked up to a FXO or FXS (it's got
2 ports) and work pretty well in production connected to a FXO port.
Don't try to bridge calls using 2 of these though. If anyone is
interested in
Hi,
I've been having a little problem with my asterisk servers, I have 4
identical asterisk servers setup (same hardware, same OS, same config). Once
in a while (once or twice a day) one of the server crashes on the cron job
reload. But I realized this only happens on 3 of the 4 servers. Tried to
Recently, I installed TDM04B 4 FXO card on to my Asterisk box and
installation went perfect.
The only problem I am facing is the Voice mail has very poor quality
when any users leave voice message via PSTN line.
We can not hear either from the extension nor from the WAV email
Dave Walker wrote:
Hi,
What could cause:
Got SIP response 406 Not Acceptable back from 10.0.0.10
10.0.0.10 = Hardware FXS
It means you have to reconfigure the device, I guess.
/O
---
Astricon 2005 - Anaheim, California, Oct 12-14 2005
http://www.astricon.net/2005/
Hello,
I have swissvoice phones and when i use one, a have in asterisk lines like:
Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp
-13691.-232125
Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp
-13691.-202125
Jul 18 17:16:22 ERROR[15251]:
Hi,
is it possible to do an attended transfer so that the original CID info will
stay for that call.
With blind transfer this works.
Regards
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I have been searching for a while and can't find
anything specific like this.
Here's is my setup:
IAXy -- broadband network
-- Asterisk -- TE110P -- Channel Bank
-- POTS lines (FXO)
Everything works fine except for the echo at the
IAXy. There is no echo on the POTS end, so Asterisk is
When I try to login into voicemail through the web interface It states
incorrect login.
In my voicemail.conf I have all voicemail boxes set under local. I
changed the symbolic
link to reflect the new directory under /var/spool/asterisk. Am I
missing something?
My vm link =
OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can
receive and place calls with no issues, however, when I receive a call, the CID
only shows Analog Line on the Grandstream 2000XP phone. Does anyone have any
ideas even where to look to change this?? Is it a setting
Does anyone know what kind of limitations asterisk has when it comes to
massive outbound dialing.. i.e. how many sip/iax phones could be dialed
at the same time-- and if someone answered, play a .wav file?
Or outbound throughput on zaptel devices?
Say if I had a dual xeon with 2 quad t1 cards,
Paul van Brouwershaven wrote:
HylaFAX can (we ar doing this now), but not with E1 or T1. So you can
only send with a maximum 2 channels. (with two default analog modems)
HylaFAX can use E1 and T1 fax modems just fine (24 and 30 channels each).
Furthermore, HylaFAX also supports multiport
Tim King a crit:
I took care of my earlier
problem. But now if I call in it just
says goodbye, And on my extension no matter what I do it seems to just
hang up
on me immediately. Its a slackware 10.1 box with Digium 22b card. I am
running AMP so its mysql driven. Im
Jian Hong GUAN wrote:
Can you tell me how to configure Hylafax + Asterisk in order to be
able to receive/send faxes.
If you have an incoming T1/E1 line:
telco -- T1/E1 -- TE405P/TE410P -- Asterisk -- TE405P/TE410P
(another port) -- T1/E1 fax modem -- HylaFAX
or:
telco -- T1/E1 --
On Mon, 2005-07-18 at 09:25 -0600, Aaron with Morad wrote:
I have been searching for a while and can't find anything specific
like this.
Here's is my setup:
IAXy -- broadband network -- Asterisk -- TE110P -- Channel
Bank -- POTS lines (FXO)
Everything works fine except for
I'm sure this topic has been discussed to death, but I haven't found a comprehensive answer yet...I have a very large installation of Cisco Call Managers connecting directly local and LD T1's for service.I would like to replace some of my LD T1's with IP trunks (or something like that). I would
HUH? Why?
If you are having Cellphones dialed for the user its one thing but
what is the goal
On 7/18/05, code select [EMAIL PROTECTED] wrote:
I'm trying to setup asterisk to accept call from Teliax, request the
10 digit number from user, then dial it thru the VoIPJet. If I'm not
wrong I
thomas DEILLON wrote:
Hello,
I have swissvoice phones and when i use one, a have in asterisk lines like:
Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp
-13691.-232125
have a idea ?
Yes, Kevin said this earlier today:
2 wrote:
i get lots of the below
I have an Asterisk setup with AMP installed. I have phone extensions
from 7000 to 7010.
I experience long delays when dialing a 9 digit number as opposed to a
10-digit number. How do you get around not having to press the # key to
speed up the dialing process? For any length phone number for
Thanks Adam. My channel banks are pretty old (NEC ND4's) so they don't do
anything for echo. I'll have to try tweaking Asterisk.
Aaron
- Original Message -
From: Adam Goryachev [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I have just purchased 20 licenses for the G729a codec from digium and set about
changing the defaults to use this codec in all cases to reduce the bandwidth
requirements (all my SIP devices support this codec). To my dismay I then find
that calls coming from SIP devices to the outside via the
I have been unable to understand the connection between an IAX
registration for dynamic IP assignment and and the host definition.
I have signed up with an ITSP for a DID. My ip is dynamic and although I
have a dynamic DNS name, we are registering and outbound works fine.
I'm at a loss to
Problem solved. Wrong context supplied by me - doh!!
Mark Phillips wrote:
I've been banging my head with this all day.
I today switched from a very old CVS build to AAH1.3 and so far
everything has been easy. However I cannot accept calls from a
previously working IAX trunk.
I've set up
Hi,
can anyone who has the Areski Calling Card solution on Asterisk
working comment on it? Is is stable enough for a production system?
Any pros and cons?
thx,
Arnd
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Hi,
-Original Message-
I have swissvoice phones and when i use one, a have in
asterisk lines like:
Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning
negative timestamp
-13691.-232125
the swissvoice firmware is IP10 SP v1.0.0 (Build 11) and
asterisk version is
the
Hi,
it is hard to answer without the right piece of extensions.conf but
remember there is a digit timeout in Asterisk: you enter 9 digits but in
the dialplan there is a match for 10 digits so how can Asterisk know if
you want to call the 9-digits number or the 10-digits? After 9 digits it
Hi Walter,
I had high load and extreme memory usage on my machine. My machine
wasn't running on SMP. My point was that the cvs version you were
using contained some bad patches, and it was probably a good idea to
upgrade or move to stable.
Thanks,
Erik
On 7/18/05, Walter Klomp [EMAIL PROTECTED]
Hi Friends,
Something I'd like to shed some light on if possible - how is it that a
single ISDN conversation only uses 64K for bidirectional communication
(using ulaw, correct?), but on several occasions now have seen
references to ulaw voip conversations using 64K per side of the
This sounds like DISA which is great for saving bucks on LD if used
right...
You will still need two channels and thus it will still cost for both
legs...
Nature of the beast...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Latham
Sent:
I have played with it. I don't know how well it would work for
production. Maybe with some custom coding etc you could get it to do
what you want. But out of the box its good for testing and nothing
more.
..o---o.
Brian Fertig
NOC/Network
Greetings,
This may be an artifact of the particular phone you are using. I know that both
Grandstream and SNOM products allow you to set a timeout for auto-dial (which
is how long to wait after the last button press before dialing the number
present).
I have my units set to three settings:
2
This is software. Use manageble software. If software means separate
setup on each desktop, then don't use it. If you spend that much time on
setting up phones, imagine how long it takes you to update other
software packages. This is, then, a symptom of a general problem.
I would like to
I have a new snom 360 on an internal net to my * box. When putting a call on
hold and taking it off, the audio will usually be broken and not
understandable.
Sometimes this happens on incoming calls and almost always on outgoing calls.
Anyone run into this before?
Thx!
--
-M
There are 10
On Monday 18 July 2005 16:04, Brian Capouch wrote:
Andrew Kohlsmith wrote:
If you don't want or don't like ABE, don't use it. Nobody is cramming it
down your throat.
I have to bite my tongue when I read these conspiratorial posts about
ABE as if it were some nefarious plot on Digium's
If you include down + up, yes, it's actually about 150-160 using uLaw +
IP/UDP/RTP/signaling overhead. But that's a little misleading, I think.
1/2 of that (~75-80) is down, 1/2 of that (~75-80) is up. So if you
have, say, a 1.5Mbps down/384 up DSL connection, you can do up to 4
calls
exten = s,9,System(curl http://127.0.0.1:13370/cgi-bin/sendsms?
username=namepassword=passto=12122122121from=12122121212text=Message
+text+here+${CALLERIDNUM})
Change it to this and it should work :
exten = s,9,System(curl http://127.0.0.1:13370/cgi-bin/sendsms?
OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can
receive and place calls with no issues, however, when I receive a call, the
CID
only shows Analog Line on the Grandstream 2000XP phone. Does anyone have
any
ideas even where to look to change this?? Is it a
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